409 Commits (8536ce8c38cd680d1a2e57d605ff6a39f7eededb)

Author	SHA1	Message	Date
Mike Bradeen	6b03d60c7d	res_monitor: Remove deprecated module. ... ASTERISK-30303 Change-Id: I0462caefb4f9544e2e2baa23c498858310b52d50	2 years ago
Mike Bradeen	e8f548c155	app_macro: Remove deprecated module. ... For most modules that interacted with app_macro, this change is limited to no longer looking for the current context from the macrocontext when set. Additionally, the following modules are impacted: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs a re-write app_queue - can no longer a macro on the called party's channel. Use gosub which is currently supported ccss - no callback macro, gosub only app_voicemail - no macro support channel - remove macrocontext and priority, no connected line or redirection macro options options - stdexten is deprecated to gosub as the default and only pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority ASTERISK-30304 Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6	2 years ago
Naveen Albert	7cbf779f13	features: Add no answer option to Bridge. ... Adds the n "no answer" option to the Bridge application so that answer supervision can not automatically be provided when Bridge is executed. Additionally, a mechanism (dialplan variable) is added to prevent bridge targets (typically the target of a masquerade) from answering the channel when they enter the bridge. ASTERISK-30223 #close Change-Id: I76f73fcd8e403bcd18f2abb40c658f537ac1ba6d	3 years ago
Naveen Albert	4a11ae7ecf	pbx: Add helper function to execute applications. ... Finding an application and executing it if found is a common task throughout Asterisk. This adds a helper function around pbx_exec to do this, to eliminate redundant code and make it easier for modules to substitute variables and execute applications by name. ASTERISK-30061 #close Change-Id: Ifee4d2825df7545fb515d763d393065675140c84	3 years ago
Alexander Traud	9440f6ec58	main: Fix for Doxygen. ... ASTERISK-29763 Change-Id: Ib8359e3590a9109eb04a5376559d040e5e21867e	3 years ago
George Joseph	647c53c41f	ACN: Changes specific to the core ... Allow passing a topology from the called channel back to the calling channel. * Added a new function ast_queue_answer() that accepts a stream topology and queues an ANSWER CONTROL frame with it as the data. This allows the called channel to indicate its resolved topology. * Added a new virtual function to the channel tech structure answer_with_stream_topology() that allows the calling channel to receive the called channel's topology. Added ast_raw_answer_with_stream_topology() that invokes that virtual function. * Modified app_dial.c and features.c to grab the topology from the ANSWER frame queued by the answering channel and send it to the calling channel with ast_raw_answer_with_stream_topology(). * Modified frame.c to automatically cleanup the reference to the topology on ANSWER frames. Added a few debugging messages to stream.c. Change-Id: I0115d2ed68d6bae0f87e85abcf16c771bdaf992c	5 years ago
George Joseph	9bd1d686a1	ACN: Add tracing to existing code ... Prior to making any modifications to the pjsip infrastructure for ACN, I've added the tracing functions to the existing code. This should make the final commit easier to review, but we can also now run a "before and after" trace. No functional changes were made with this commit. Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c	5 years ago
Richard Mudgett	0376f2bba9	features.c: Make Bridge application tolerate unspecified channel. ... The Bridge application was inconsistent if the channel to bridge with is not specified. If no parameters are given then a warning is issued and the current channel is hung up. If options are given but no channel is specified then a warning is issued and the current channel is not hung up. * Made the Bridge application give a verbose message instead of a warning if the channel to bridge with is not specified and made not hang up the current channel. As a result dialplan no longer needs to check if a channel name is passed before calling Bridge and simply needs to check the BRIDGERESULT channel variable instead. This is something you likely want your dialplan to do anyway. * Fixed up L() option warning message. It is up to the caller to determine if the channel is hung up because of the warning. Dial() hangs up the current channel while Bridge() does not. Change-Id: I44349a8dc3912397f28852777de04f19e7bb9c73	5 years ago
Corey Farrell	709f4b81e7	loader: Process dependencies for built-in modules. ... With the new module loader it was missed that built-in modules never parsed dependencies from mod->info into vectors of mod. This caused manager to be initialized before acl (named_acl). If manager.conf used any named ACL's they would not be found and result in no ACL being applied to the AMI user. In addition to the manager ACL fix this adds "extconfig" to all builtin modules which support realtime configuration. This only matters if one of the builtin modules is configured with 'preload', depending on "extconfig" will cause config.c to automatically be initialize during the preload stage. Change-Id: I482ed6bca6c1064b05bb538d7861cd7a4f02d9fc	7 years ago
Corey Farrell	572a508ef2	loader: Convert reload_classes to built-in modules. ... * acl (named_acl.c) * cdr * cel * ccss * dnsmgr * dsp * enum * extconfig (config.c) * features * http * indications * logger * manager * plc * sounds * udptl These modules are now loaded at appropriate time by the module loader. Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so the module loader will abort startup on failure of these modules. Some of these modules are still initialized or shutdown from outside the module loader. logger.c is initialized very early and shutdown very late, manager.c is initialized by the module loader but is shutdown by the Asterisk core (too much uses it without holding references). Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f	7 years ago
Corey Farrell	fee929c8ac	core: Remove non-critical cleanup from startup aborts. ... When built-in components of Asterisk fail to start they cause the Asterisk startup to abort. In these cases only the most critical cleanup should be performed - closing databases and terminating proceses. These cleanups are registered using ast_register_atexit, all other cleanups should not be run during startup abort. The main reason for this change is that these cleanup procedures are untestable from the partially initialized states, if they fail it could prevent us from ever running the critical cleanup with ast_run_atexits. Create separate initialization for dns_core.c to be run unconditionally during startup instead of being initialized by the first dns resolver to be registered. This ensures that 'sched' is initialized before it can be potentially used. Replace ast_register_atexit with ast_register_cleanup in media_cache.c. There is no reason for this cleanup to happen unconditionally. Change-Id: Iecc2df98008b21509925ff16740bd5fa29527db3	7 years ago
Kevin Harwell	1e70011710	features: Bridge application's BRIDGERESULT not appropriately set ... The dialplan application "Bridge" was not setting the BRIDGERESULT to failure when a failure did occur. Even worse if it did fail to join the bridge it would still report success. This patch now sets the BRIDGERESULT variable to an appropriate value for a given condition state. Also, removed the value INCOMPATIBLE as a valid result type since it is no longer used. ASTERISK-27369 #close Change-Id: I22588e7125a765edf35cff28c98ca143e9927554	8 years ago
Joshua Colp	c90d81ef51	bridge: Fix returning to dialplan when executing Bridge() from AMI. ... When using the Bridge AMI action on the same channel multiple times it was possible for the channel to return to the wrong location in the dialplan if the other party hung up. This happened because the priority of the channel was not preserved across each action invocation and it would fail to move on to the next priority in other cases. This change makes it so that the priority of a channel is preserved when taking control of it from another thread and it is incremented as appropriate such that the priority reflects where the channel should next be executed in the dialplan, not where it may or may not currently be. The Bridge AMI action was also changed to ensure that it too starts the channels at the next location in the dialplan. ASTERISK-24529 Change-Id: I52406669cf64208aef7252a65b63ade31fbf7a5a	8 years ago
Corey Farrell	a6e5bae3ef	Remove ASTERISK_REGISTER_FILE. ... ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966	9 years ago
Matt Jordan	a93cd39ac1	manager: Add <see-also> tags to relate Bridge related events,actions, and apps ... Change-Id: I67e6b79fa3102e494b5fe6cc7510472249080e85	9 years ago
Corey Farrell	d3348c51b5	features.c: Remove unneeded adsi.h include. ... adsi.h is no longer used by features.c since parking was moved to a module. Change-Id: I2248b8a455225a17cb6ddaafd6c20c511a1eaf59	9 years ago
Richard Mudgett	4f7b859726	features: Fix channel datastore access. ... Found as a result of the testsuite tests/callparking test crashing. Several calls to ast_get_chan_featuremap_config() and ast_get_chan_features_xfer_config() did not lock the channel before calling so the channel's datastore list was accessed without the lock's protection. Apparently another thread deleted a datastore on the channel's list while the crashing thread was walking the list. Crash at 0xdeaddead due to MALLOC_DEBUG's memory filler value as a result. * Add missing channel locks to calls that were not already protected as the doxygen for those calls indicates. Change-Id: Id273b3d305cc616406c353cbc841b2b7655efaa1	9 years ago
Timo Teräs	39b69ab537	Fixes to include signal.h ... POSIX defines signal.h. sys/signal.h should not be used as it is c-library internal header which may or may not exist. Notably with musl it generates warning of being incorrect. Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc	9 years ago
Richard Mudgett	a63656b419	Bridge system: Fix memory leaks and double frees on impart failure. ... You cannot reference the passed in features struct after calling ast_bridge_impart(). Even if the call fails. Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21	9 years ago
Corey Farrell	c3ec5da156	Remove unneeded uses of optional_api providers. ... A few cases exist where headers of optional_api provders are included but not needed. This causes unneeded calls to ast_optional_api_use. * Don't include optional_api.h from sip_api.h. * Move 'struct ast_channel_monitor' to channel.h. * Don't include monitor.h from chan_sip.c, channel.c or features.c. The move of struct ast_channel_monitor is needed since channel.c depends on it. This has no effect on users of monitor.h since channel.h is included from monitor.h. ASTERISK-25051 #close Reported by: Corey Farrell Change-Id: I53ea65a9fc9693c89f8bcfd6120649bfcfbc3478	10 years ago
Mark Michelson	aae45acbda	Detect potential forwarding loops based on count. ... A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23	10 years ago
Matt Jordan	4a58261694	git migration: Refactor the ASTERISK_FILE_VERSION macro ... Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e	10 years ago
Matthew Jordan	d2776d4d45	clang compiler warnings: Fix a variety of "unused" warnings ... This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable errors caught by clang. Specifically: * apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[], qsmp_cmd_usage[] * cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom" * channels/chan_pjsip.c: removed unused desc[] = "PJSIP Channel" * codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$" * funcs/func_env.c:729: Fixed ast_str_append_substr. * main/editline/np/strlcat.c: removed unused rcsid variable * main/editline/np/strlcpy.c: removed unused rcsid variable * main/security_events.c: removed unused TIMESTAMP_STR_LEN * utils/conf2ael.c: removed unused cfextension_states * utils/extconf.c: removed unused cfextension_states Review: https://reviewboard.asterisk.org/r/4526 ASTERISK-24917 Reported by: dkdegroot patches: rb4526.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433693 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433694 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433695 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	3ddd92902a	Replace most uses of ast_register_atexit with ast_register_cleanup. ... Since 'core stop now' and 'core restart now' do not stop modules, it is unsafe for most of the core to run cleanups. Originally all cleanups used ast_register_atexit, and were only changed when it was shown to be unsafe. ast_register_atexit is now used only when absolutely required to prevent corruption and close child processes. Exceptions that need to use ast_register_atexit: * CDR: Flush records. * res_musiconhold: Kill external applications. * AstDB: Close the DB. * canary_exit: Kill canary process. ASTERISK-24142 #close Reported by: David Brillert ASTERISK-24683 #close Reported by: Peter Katzmann ASTERISK-24805 #close Reported by: Badalian Vyacheslav ASTERISK-24881 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4500/ Review: https://reviewboard.asterisk.org/r/4501/ ........ Merged revisions 433495 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433497 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433498 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	c08fd275bf	Logger: Convert 'struct ast_callid' to unsigned int. ... Switch logger callid's from AO2 objects to simple integers. This helps in two ways. Copying integers is faster than referencing AO2 objects, so this will result in a small reduction in logger overhead. This also erases the possibility of an infinate loop caused by an invalid callid in threadstorage. ASTERISK-24833 #comment Committed callid conversion to trunk. Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4466/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Scott Griepentrog	5b30938394	app_bridge: return to the next dialplan priority ... When app_bridge grabs a channel and puts it into a bridge, the channel should then continue where it left off in the dialplan after the bridge has ended. Although it stores the current dialplan location as an after bridge goto on the channel, it was executing the same priority again instead of going to the next priority. By swapping the "specific" version of bridge_set_after_goto with bridge_set_after_go_on, the next priority in the dialplan is executed instead. ASTERISK-24637 #close Review: https://reviewboard.asterisk.org/r/4322/ Reported by: John Bigelow ........ Merged revisions 430467 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	ec1a7654f3	Fix leak in AMI Action Bridge ... Add missing reference cleanup for newly created bridge. ASTERISK-24281 Reported by: Stefan Engström Review: https://reviewboard.asterisk.org/r/4154/ ........ Merged revisions 427736 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427737 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c384532aa4	features.c: Fix lingering channel ref while Bridge() application is active. ... Using the Bridge application to bridge a channel that is executing an applicaiton such as Wait results in a lingering Surrogate channel in the CLI "core show channels" output even though it has already hungup. * Fix bridge_exec() to not hold onto the current_dest_chan ref once it has been put into the bridge. * Eliminated bridge_exec()'s use of RAII_VAR(). ASTERISK-24224 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4041/ ........ Merged revisions 424668 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424669 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	f6647d2362	Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027 ... Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue. Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample. (issue ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046) (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine (license 6561) hyphen.patch uploaded by Jeremy Laine (license 6561) sip.conf.sample.patch uploaded by Eugene (license 6360) ........ Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405792 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405829 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2a371cd80b	Restore Dial, Queue, and FollowMe 'I' option support. ... The Dial, Queue, and FollowMe applications need to inhibit the bridging initial connected line exchange in order to support the 'I' option. * Replaced the pass_reference flag on ast_bridge_join() with a flags parameter to pass other flags defined by enum ast_bridge_join_flags. * Replaced the independent flag on ast_bridge_impart() with a flags parameter to pass other flags defined by enum ast_bridge_impart_flags. * Since the Dial, Queue, and FollowMe applications are now the only callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the calling contract to require the initial COLP exchange to already have been done by the caller. * Made all callers of ast_bridge_impart() check the return value. It is important. As a precaution, I also made the compiler complain now if it is not checked. * Did some cleanup in parking_tests.c as a result of checking the ast_bridge_impart() return value. An independent, but associated change is: * Reduce stack usage in ast_indicate_data() and add a dropping redundant connected line verbose message. (closes issue ASTERISK-22072) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/2845/ ........ Merged revisions 399136 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	477dea4661	Bridge API: Set a cause code on a channel when it is ejected from a bridge. ... The cause code needs to be passed from the disconnecting channel to the bridge peers if the disconnecting channel dissolves the bridge. * Made the call to an app_agent_pool agent disconnect with the busy cause code if the agent does not ack the call in time or hangs up before acking the call. (closes issue ASTERISK-22042) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2772/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	00baddb906	Massively clean up app_queue. ... This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b816fe45b6	* Move ast_bridge_channel_setup_features() into bridge_basic.c. ... * Made application map hooks be removed on a basic bridge personality change. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d213dfa30f	Fix several interrelated issues dealing with the holding bridge technology. ... * Added an option flags parameter to interval hooks. Interval hooks now can specify if the callback will affect the media path or not. * Added an option flags parameter to the bridge action custom callback. The action callback now can specify if the callback will affect the media path or not. * Made the holding bridge technology reexamine the participant idle mode option whenever the entertainment is restarted. * Fixed app_agent_pool waiting agents needlessly starting and stopping MOH every second by specifying the heartbeat interval hook as not affecting the media path. * Fixed app_agent_pool agent alert from restarting the MOH after the alert beep. The agent entertainment is now changed from MOH to silence after the alert beep. * Fixed holding bridge technology to defer starting the entertainment. It was previously a mixture of immediate and deferred. * Fixed holding bridge technology to immediately stop the entertainment. It was previously a mixture of immediate and deferred. If the channel left the bridging system, any deferred stopping was discarded before taking effect. * Miscellaneous holding bridge technology rework coding improvements. Review: https://reviewboard.asterisk.org/r/2761/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e47d3db365	Doxygen comment tweaks. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	8b7742202f	Minor parking cleanup. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396812 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	58af87ef2c	Remove early bridge BUGBUG comments. Remove some unneeded features.c comments. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c3466db29d	Resolve some BUGBUG comments. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	6d24165dee	Remove some dead code dealing with: AST_BRIDGE_REC_CHANNEL_0, AST_BRIDGE_REC_CHANNEL_1, and AST_BRIDGE_IGNORE_SIGS. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	82ba10bb47	Fix feature_attended_transfer test ... The feature_attended_transfer test is failing due to Asterisk not passing DTMF in the bridges created for internal attended transfers. This sets the features initialization routine to set this flag by default and adjusts the basic bridge and confbridge's use of the bridging system accordingly as per Richard's suggestion instead of adjusting this individual case. This change allows the necessary DTMF to pass through the attended transfer bridge and complete the test successfully. Review: https://reviewboard.asterisk.org/r/2759/ (closes issue ASTERISK-22222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	63b5bf26ec	Fix two race conditions and ref counting issue when joining a bridge ... These problems were all caught by a test in the Asterisk Test Suite that originated some Local channels and attempted to move the ;2 half of the Local channel into a bridge using the Bridge AMI action. (1) When originating a channel, the Newchannel event is emitted quickly; however, the ;2 channel will not have a pbx thread assigned to it until after the outbound 'dialing' for the ;1 is complete. Thus, there is a period of time where the outside world "knows" of the channel's existence and can influence it but Asterisk has not yet started the dialplan execution thread. If a Bridge AMI action is taken on the channel, the channel appears to be a Dialed channel with no PBX thread; hence, the channel will be imparted into the Bridge by first 'yanking' the channel. At the same time, a race condition can occur after the yank (but before entering the bridge) when ;1 answers and starts a PBX on the ;2. The end result currently is an assertion failure in the Bridging API, as a channel with a PBX is imparted into the Bridge. There's no way to prevent AMI from attempting to Bridge a channel immediately after creation; likewise, holding the channel lock through the entire Dial operation is unwise (and impossible). Instead of treating the presence of a PBX thread as an error, we simply bail out of the adding the channel to the bridge through ast_bridge_impart. The Bridge action will then fail - but we avoid a situation where the channel is both executing a PBX thread and simultaneously being given a separate thread in the bridging system (which would be a "bad thing"). Since imparting a channel with a PBX *can* occur and is not a programming error, the asserts have been removed. (2) When the first condition occurs, we have to take one of two actions: either hangup the yanked channel as it did not enter the bridge, or deref it because we don't own it. We can determine if we own it or not by testing for the presence of the PBX thread. If we hung it up directly, we'd crash. (3) bridge_find_channel does not increase the reference count of the ast_bridge_channel object. The RAII_VAR usage in ast_bridge_add_channel thus created a ticking time bomb in whatever bridge the channel moved into, as the destructor for the ast_bridge_channel object would be called. Review: https://reviewboard.asterisk.org/r/2741/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	73b3c70a5f	Remove some resolved or obsolete BUGBUG comments. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396401 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	3797639e84	bridge features: Dial and Queue add features instead of replace them. ... Dial and Queue would previously apply a new set of features whenever bridging. These options would be based purely on the options supplied to the dial/queue applications. This patch changes the function those applications use to bridge calls so that the features will be added to the set of existing features for each channel rather than having them override the existing features. (closes issue ASTERISK-22209) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/2713/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	38236e54a8	Remove dead code from features.c; refactor pickup code into pickup.c ... This patch does the following: * It moves the pickup code out of features.c and into pickup.c * It removes the vast majority of dead code out of features.c. In particular, this includes the parking code. (issue ASTERISK-22134) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	5c4b482471	Support externally initiated parking requests; remove some dead code ... This patch does the following: * It adds support for externally initiated parking requests. In particular, chan_skinny has a protocol level message that initiates a call park. This patch now supports that option, as well as the protocol specific mechanisms in chan_dahdi/sig_analog and chan_mgcp. * A parking bridge features virtual table has been added that provides access to the parking functionality that the Bridging API needs. This includes requests to park an entire 'call' (with little or no additional information, thank you chan_skinny), perform a blind transfer to a parking extension, determine if an extension is a parking extension, as well as the actual "do the parking" request from the Bridging API. * Refactoring in chan_mgcp, chan_skinny, and chan_dahdi to make use of the new functions * The removal of some - but not all - dead parking code from features.c This also fixed blind transferring a multi-party bridge to a parking lot (which was implemented, but had at least one code path where using the parking features kK might not have worked) Review: https://reviewboard.asterisk.org/r/2710 (closes issue ASTERISK-22134) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	03090a88ba	Fix documentation replication issues ... This prevents XML documentation duplication by expanding channel and bridge snapshot tags into channel and bridge snapshot parameter sets with a given prefix or defaulting to no prefix. This also prevents documentation from becoming fractured and out of date by keeping all variations of the documentation in template form such that it only needs to be updated once and keeps maintenance to a minimum. Review: https://reviewboard.asterisk.org/r/2708/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c017d5e6a3	Remove the unsafe bridge parameter from ast_bridge_hook_callback's. ... Most hook callbacks did not need the bridge parameter. The pointer value could become invalid if the channel is moved to another bridge while it is executing. * Fixed some issues in feature_attended_transfer() as a result. * Reduce the bridge inhibit count in attended_transfer_properties_shutdown() after it has restored the bridge channel hooks. * Removed basic bridge requirement on feature_blind_transfer(). It does not require the basic bridge like feature_attended_transfer(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395574 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	93a70d83e3	Remove some dead parking call ... Since nothing is using these global parking functions, remove them! The first of many. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395410 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	fbcc3addf8	Remove dead bridging code from features ... This removes the previously #if 0'd code. The functionality removed has either been subsumed by the Bridging API or is no longer applicable. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	cafc115896	A great big renaming patch ... This patch renames the bridging* files to bridge*. This may seem pedantic and silly, but it fits better in line with current Asterisk naming conventions: * channel is not "channeling" * monitor is not "monitoring" etc. A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is the act of using a bridge on a set of channels - and the API that fulfills that role is more than just the action. (closes issue ASTERISK-22130) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago